(old)Puppy Linux Discussion Forum

Hi Will,

My request is for a "ringin" sound when a call comes in. That way if someone is away from the keyboard an alert is sounded to let us know a call is pending. Other than that it's a keeper and runs well here.

Best,
Eric

add your feature requests here

Conference calling
One voice mail to many recipients
Video well OK we need to recompile the webcam driver
wizard from 2.xx to 4.xx

. . . now I am thinking something rather different
a message server - anyone up for this?

1. Leave reviews and feedback
2. Report bugs
3. Request help
4. Tutorial

How would this work? Well . . for example . . .
Assuming we have server space
We have a menu option called Puppy
with the above options
Each of the above options would have two variables
an email address and the name of the sender
eg puppyhelp@gmail.com (with redirect)

We are Puppy. We bark.

Here are my requests. Not sure if these are already in the latest version as I did not get a chance to test that. Cannot run Puppy from office.

- IM window more professional like Gizmo, Skype etc.
- As Eric (Caneri) said, ringing sound when a call comes in.
- When closing the application, the application minimizes to tray or quick launch location and can be restored back.
- Main first window has buddy list and status.

Basically, I am dreaming of seeing PSip work similar to the big players like Gizmo, Skype etc. If this is too much to ask, please pardon me.

Installing 0.9.12 now, and would really love to have someone to test it out. Is there anyone that will be online at some point to give me a person to chat with?

gaz and feverfew
thanks for the crash reports
the packet loss doesn't look too high, normally there are about 50 packets a second transmitted.
The jitter is very high. If you both have reasonable CPUs you could edit the config file to up the quality to 10.
experimenting with the
--no-vad
option might help as well
gaz please can you edit your config to
--log-file /tmp/psip/app-log
so that it gets included in the log

"gaz please can you edit your config to
--log-file /tmp/psip/app-log
so that it gets included in the log"


That's Strange... I thought we went over that , He made sure that app.log was in the tar.gz

Was mine missing to?

psip-0.9.13
hot off the press
buddy list in the main window (highlight your buddy and hit call or im)
no popups for presence announcements there is a dynamic infobar (thanks zigbert for the inspiration)
automatically sets your status to On the Phone and Online (when you hang up)
The IM entry window submits automatically when you hit return.

have fun

if upgrading make sure that your config file contains
--log-file /tmp/psip/app-log

Feverfew wrote:Was mine missing to?

yours was fine
gary's was there but empty

Here is my report from 0.9.12

Good
- I could listen to all my voice messages without interruption. I was not able to in the previous versions.

Not good
- I tried calling Lobster and could hear his voice, but he could not hear mine even though my mic was working. I think earlier version 0.9.8 did work well with my mic.
- On subsequent attempts to call, I no longer was hearing a dialing tone at all. All of a sudden, I could hear Lobster speaking.

As I am not on my Puppy system when posting this, I am unable to send a dump.

I see that the buddy list is on the main window in 0.9.13. This is wonderful. Thanks Will.

Hi Will,

First impressions are GREAT...

I just had to reload the buddy list to populate it...other than that, seems ok...I'll check it out today as I go along.
Thanks,
Eric

buddy list in the main window (highlight your buddy and hit call or im)

much better - can we move it to left of icons for eeepc use?
and eh . . record voice conversations?

strange but i do not know why i am not heard....
i use a first gui version of psip
and it worked fine before today
i did not do anything with my microfone but the person on the other side do not hear me

The monkeying about that lobster, smokey and I had to do to get sound working has highlighted the sound interface as a major source of aggravation.

When building psip-0.9.12 I made a major change to the underlying portaudio sound architecture and I don't really appreciate the implications.

If anyone finds that psip-0.9.-11 works and latter versions don't then I may need to consider providing parallel builds using different versions of portaudio. It is pretty hard for me to debug these problems as my sound system works with either. When running pjsua in cli mode it should be possible to investigate what sound interface it is using. Anyone have audio problems is advised to check out the pjsip project links on the help menu. It is fairly heavy going stuff but I can't do it all from this end.

Wait... 0.9.13 is already out? Wow you guys move fast.

Quick questions: Are you able to speak to Skype members from this? Would it be free? And also, if there is any type of pay involved, which features are free?

Speaking to any otehr Gizmo/PSIP user is free. Calling a landline or cellphone I believe costs whatever Gizmo charges. Chatting is free as is calling the members in the buddy list. I'm not 100% sure on this, but it's what I've gathered from my own experimentation.

Evil20071 wrote:Speaking to any otehr Gizmo/PSIP user is free. Calling a landline or cellphone I believe costs whatever Gizmo charges. Chatting is free as is calling the members in the buddy list. I'm not 100% sure on this, but it's what I've gathered from my own experimentation.

So is Skype part of the Gizmo/PSIP by any chance?

exProphecy wrote:So is Skype part of the Gizmo/PSIP by any chance?

Skype is closed source and a private protocol. If you want to talk to skype users you have to use skype, unless someone has reverse engineered their protocol.

Feverfew wrote:"gaz please can you edit your config to
--log-file /tmp/psip/app-log
so that it gets included in the log"


That's Strange... I thought we went over that , He made sure that app.log was in the tar.gz

Was mine missing to?

I have now put this in my config file..didnt have it before

I'm seriously liking the new layout, especially the online/offline bar under the dialpad. I also noticed that, while playing around with the configuration file, that if you add your own user id to the buddy list section of it, it crashes PSIP.